Audio signal processing apparatus, audio signal processing method, and communication terminal

ABSTRACT

An audio signal processing apparatus includes a band signal component extraction unit for extracting signal components in a predetermined frequency band from an audio signal input by an audio collection unit for collecting at least an outgoing call audio upon an audio phone call, a stationary signal component extraction unit for extracting a stationary signal component from the signal components, a signal adjustment unit having a level adjustment function of adjusting an output signal level with respect to an input signal level and an input and output characteristics change function of changing input and output characteristics upon level adjustment in the level adjustment function through a control signal and configured to set an incoming call audio signal upon the audio phone call as the input signal, and a control signal generation unit for generating the control signal for changing the input and output characteristics by using the stationary signal component.

BACKGROUND OF THE INVENTION

1. Field of the Invention

The present invention relates to an audio signal processing apparatusand an audio signal processing method for making it easier to hear avoice on an incoming call sent via a communication network such as amobile phone network and a communication terminal such as a mobile phoneterminal enabling an audio phone call.

2. Description of the Related Art

Up to now, for a case in which an audio phone call is carried out via acommunication network such as a mobile phone network, for example, atechnology is proposed for improving the ease of hearing a voice on thephone call in an environment with an ambient noise by applying apredetermined signal processing on a phone call audio signal on an audioreception side.

For example, Japanese Unexamined Patent Application Publication No.7-221832 (FIG. 1) discloses a technology for improving the ease ofhearing by comparing frequency characteristics of an ambient noise andan incoming call audio and changing the frequency characteristic of theincoming call audio.

SUMMARY OF THE INVENTION

However, as described above, in order to compare the frequencycharacteristics of the ambient noise and the incoming call audio andchange the frequency characteristic of the incoming call audio, a largenumber of processing are carried out such as a signal analysis on boththe ambient noise and the incoming call audio and a comparison betweenthese analysis results. For this reason, for example, there is a problemthat burden becomes large for an apparatus having a limited processingperformance such as a mobile phone terminal.

The mobile phone terminal is used in various environments in general.Among the use environments, in particular, in a case where the mobilephone terminal is used in an environment such as a party venue or a pub,voice from other people who surround a talker having a phone callbecomes a source of noise. However, a technology for making it easier tohear the incoming call audio has not existed yet in the use of themobile phone terminal under the environment where the noise from theother people exists in this way.

The present invention has been proposed in view of the above-mentionedcircumstances, and it is desirable to provide an audio signal processingapparatus, an audio signal processing method, and a communicationterminal in which it is possible to make it easier to hear a phone callaudio (voice on an incoming call) with a small processing amount, inparticular, an audio signal processing apparatus, an audio signalprocessing method, and a communication terminal in which it is possibleto make it easier to hear the voice on the phone call even under theenvironment where voice from other people other than a talker having aphone call becomes a source of noise.

According to an embodiment of the present invention, there is providedan audio signal processing apparatus including: a band signal componentextraction unit configured to extract signal components in apredetermined frequency band from an audio signal input by audiocollection means configured to collect at least an outgoing call audioupon an audio phone call; a stationary signal component extraction unitconfigured to extract at least a stationary signal component from thesignal components extracted by the band signal component extractionunit; a signal adjustment unit provided with a level adjustment functionof adjusting an output signal level with respect to an input signallevel and an input and output characteristics change function ofchanging input and output characteristics upon level adjustment in thelevel adjustment function through a control signal and configured to setan incoming call audio signal upon the audio phone call as the inputsignal; and a control signal generation unit configured to generate thecontrol signal for changing the input and output characteristics of thesignal adjustment unit by using at least the stationary signal componentextracted by the stationary signal component extraction unit.

That is, according to the embodiment of the present invention, thesignal components in the predetermined frequency band are extracted fromthe audio signal input by the audio collection means. For thepredetermined frequency band, the frequency band of the human voice canbe exemplified. Among the signal components in the predeterminedfrequency band, in particular, the stationary signal component can beconsidered as the signal component of the voice from surrounding otherpeople except for the voice of the talker having the phone call.Therefore, according to the embodiment of the present invention, as thestationary signal component is used, the level of the incoming callaudio signal upon the audio phone call is adjusted. To be more specific,in accordance with the size of the level of the signal component, thedynamics on the audio reception side is controlled.

In addition, according to a further embodiment, in the audio signalprocessing apparatus, the band signal component extraction unit mayinclude a first filter designed to extract a signal waveform in thefirst frequency band from an input audio signal, a second filterdesigned to extract a signal waveform in the second frequency band fromthe input audio signal, a first envelope detector configured to detectan envelope of a signal waveform after pass of the first filter, and asecond envelope detector configured to detect an envelope of a signalwaveform after pass of the second filter, output a signal waveform afterthe envelop detection by the first envelope detector as the signalcomponents in the first frequency band, and output a signal waveformafter the envelop detection by the second envelope detector as thesignal components in the second frequency band, the stationary signalcomponent extraction unit may include a mute unit configured to mute thesignal waveform after the envelop detection by the first envelopedetector and a mute control unit configured to cancel mute of the muteunit when a signal level of the signal waveform after the envelopdetection exceeds a predetermined threshold and also this state carrieson for a predetermined period of time and to enable the mute of the muteunit when the signal level dips from the predetermined threshold afterthe mute cancellation and output an output waveform of the mute unit asthe stationary signal component, and the control signal generation unitmay use the stationary signal component composed of a signal waveformoutput from the mute unit and the signal components in the secondfrequency band composed of the signal waveform after the envelopdetection by the second envelope detector to generate the controlsignal.

That is, according to the embodiment of the present invention, as thesignal components in the predetermined frequency band, the signalcomponents in the first frequency band and the second frequency band areextracted. For the first frequency band, the frequency band of the humanvoice can be exemplified. For the second frequency band, a frequencyband lower than the frequency band of the human voice can beexemplified. Then, according to the embodiment of the present invention,the control signal is generated on the basis of the signal componentshaving passed the mute unit among the signal components in the firstfrequency band and the signal components in the second frequency band.

According to the embodiment of the present invention, on the basis ofthe signal components in the predetermined frequency band extracted fromthe input audio signal, the level of the incoming call audio signal uponthe audio phone call is adjusted. That is, for example, when the ambientenvironment noise is large, a level adjustment is carried out in amanner that the level of the incoming call audio signal is increased.Thus, it is possible to make it easier to hear the phone call audio (inparticular, the voice on the incoming call) with a small processingamount. In particular, according to the embodiment of the presentinvention, the frequency band of the human voice is used for thepredetermined frequency band. Thus, at the time of the call under theenvironment where the human voice is included as the noise source, it ispossible to make it easier to hear the incoming call audio.

In addition, for the predetermined frequency band, for example, thefirst frequency band of the human voice and the second frequency bandwhich is lower than the first frequency band are respectively extracted,and temporal delays used for analysis in the respective bands areminimized and combined with each other, so that it is possible to applyto the ambient environment noise in a wide band, and also the leveladjustment of the incoming call audio signal at a faster response timecan be realized.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a block diagram of a schematic configuration of a mobile phoneterminal according to an embodiment of the present invention;

FIG. 2 is a block circuit diagram of detailed configurations of a noisedetection unit and an incoming call audio processing unit provided in anaudio processing unit of the mobile phone terminal according to thepresent embodiment;

FIG. 3 is a schematic diagram used for describing a frequencycharacteristic of a human voice;

FIG. 4 is an amplitude-frequency characteristic diagram of an actuallymeasured incoming call audio signal including an ambient environmentnoise;

FIG. 5 is an amplitude-frequency characteristic diagram of the incomingcall audio signal of FIG. 4 in which 0 Hz to 300 Hz are expanded;

FIG. 6 is an amplitude-frequency characteristic diagram of a formant ofthe actually measured incoming call audio signal;

FIG. 7 is an amplitude-frequency characteristic diagram of the incomingcall audio signal of FIG. 4 in which 0 Hz to 300 Hz are expanded;

FIG. 8 is a characteristic diagram of a frequency characteristic of alow-pass filter in the noise detection unit;

FIG. 9 is a characteristic diagram in which a low frequency band part isexpanded in particular among the frequency characteristic of thelow-pass filter in the noise detection unit;

FIG. 10 is a waveform diagram of an audio signal waveform input from amicrophone and a post LPF pass signal waveform obtained from the audiosignal waveform after passing through the low-pass filter in the noisedetection unit;

FIG. 11 is a characteristic diagram of a frequency characteristic of aband-pass filter in the noise detection unit;

FIG. 12 is a waveform diagram of an audio signal waveform input from amicrophone and a past BPF pass signal waveform obtained by from theaudio signal waveform after passing through the band-pass filter in thenoise detection unit;

FIGS. 13A to 13C are waveform diagrams of an output signal waveformexample of an envelope detector on a branch path side of the band-passfilter, an output signal waveform example of a mute circuit, and anoutput signal waveform example of a mixing volume circuit;

FIGS. 14A to 14C are waveform diagrams of an output signal waveformexample of an envelope detector on a branch path side of the low-passfilter, an output signal waveform example of the mixing volume circuiton the branch path side of the band-pass filter, and an output signalwaveform example of a comparator;

FIG. 15 is a waveform diagram of a waveform example of a control signal(a signal representing a rough energy transient of the ambientenvironment noise) output from the noise detection unit;

FIG. 16 is a waveform diagram of a waveform example in which the controlsignal output from the noise detection unit passes through a limiter andan amplifier of a dynamics adjustment unit;

FIG. 17 is a characteristic diagram of input and output characteristiccurves of an auto level controller having a variable hinge pointaccording to the present embodiment;

FIG. 18 is a characteristic diagram in which a vicinity of the variablehinge point of FIG. 17 is expanded;

FIG. 19 is a diagram used for describing a relation between a levelchange of the control signal and a change of the variable hinge point;

FIG. 20 is a waveform diagram of a schematic amplitude waveform of anincoming call audio signal after the level adjustment by the auto levelcontroller according to the present embodiment;

FIG. 21 is a characteristic diagram used for describing an example inwhich the input and output characteristics are changed in a state ofhaving a line segment in parallel with the input and outputcharacteristic curves of the auto level controller in which the inputlevel and the output level correspond to one to one;

FIG. 22 shows an example of a relation between the control signal andthe hinge point in a case where the input and output characteristics ofthe auto level controller are controlled through a digital processing;

FIG. 23 shows another example of a relation between the control signaland the hinge point in a case where the input and output characteristicsof the auto level controller are controlled through the digitalprocessing;

FIG. 24 is a waveform diagram of a signal waveform after the controlsignal of FIG. 15 output from the noise detection unit passes through alimiter and an amplifier of a formant adjustment unit;

FIG. 25 is a characteristic diagram of a frequency characteristic of aband-pass filter of the formant adjustment unit;

FIG. 26 is an amplitude-frequency measurement diagram obtained byactually measuring the incoming call audio signal input from an incomingcall audio signal input terminal;

FIG. 27 is an amplitude-frequency measurement diagram obtained byactually measuring the incoming call audio signal after a band passprocessing by the band-pass filter of the formant adjustment unit;

FIG. 28 is a characteristic diagram of a relation between the controlsignal in the amplifier of the formant adjustment unit and anamplification factor;

FIG. 29 is a schematic diagram used for describing a state in which thefrequency characteristic of the second formant in the frequencycharacteristic of the human voice is adjusted by the formant adjustmentunit;

FIG. 30 is an amplitude-frequency measurement diagram obtained byactually measuring the incoming call audio signal after a gainadjustment by the amplifier of the formant adjustment unit;

FIG. 31 is an amplitude-frequency measurement diagram obtained byactually measuring the incoming call audio signal after an additionprocessing by an adder of the formant adjustment unit; and

FIG. 32 is a block diagram used for describing another schematicconfiguration example of the noise detection unit provided in the audioprocessing unit of the mobile phone terminal according to the presentembodiment.

DESCRIPTION OF THE PREFERRED EMBODIMENTS

Hereinafter, with reference to the drawings, an embodiment of thepresent invention will be described.

It should be noted that according to the present embodiment, a mobilephone terminal is described as an example of the present invention, butof course, a content described herein are merely an example, and thepresent invention is not limited to this example for sure.

Schematic Configuration of the Mobile Phone Terminal

FIG. 1 shows a schematic configuration of the mobile phone terminalaccording to the present embodiment.

In FIG. 1, a communication antenna 12 is, for example, a built-inantenna and carries out a transmission of signal waves for a phone callor a packet communication such as an electronic mail. A communicationcircuit 11 performs a frequency conversion, a modulation, ademodulation, and the like for a transmission signal.

A control unit 10 is composed of a CPU (central processing unit) andperforms a control on a communication in the communication circuit 11, acontrol on an audio processing, a control on an image processing,various other signal processings, controls on the respective units, andthe like. Also, the control unit 10 performs an execution on variouscontrol programs and application programs accumulated in a memory unit15, various data processings accompanying the execution, and the like.

A speaker 21 is composed of a speaker for an incoming call provided inthe mobile phone terminal and a speaker for outputting a ringer (ringalert), an alarm sound, a warning sound, a reproduced music, a digitalaudio, an audio of a reproduced video, and the like. The speaker 21converts the audio signal supplied from an audio processing unit 20 intoan acoustical wave to be output into the air.

A microphone 22 is a microphone for collecting an outgoing call and anexternal audio and is configured to convert an acoustical wave into anaudio signal and input the audio signal to the audio processing unit 20.

The audio processing unit 20 amplifies an audio data generated through apredetermined audio processing such as the demodulation afterdigital/analog conversion and outputs the audio signal after theamplification to the speaker 21. Also, the audio processing unit 20performs amplification and analog/digital conversion on the input audiosignal supplied from the microphone 22 and applies a predetermined audioprocessing such as encoding on the audio data after the analog/digitalconversion. Also, in particular, in a case where the mobile phoneterminal according to the embodiment of the present invention, the audioprocessing unit 20 is provided with a noise detection unit 23 and anincoming call audio processing unit 24 are provided. Detailedconfigurations and operations of the noise detection unit 23 and theincoming call audio processing unit 24 will be described below.

An operation unit 13 is composed of various operation elements includingvarious keys such as numeric keypads, a call key, and a call end/powerkey and various operation elements such as an arrow key and a jog dialwhich are provided on an enclosure (not shown) of the mobile phoneterminal according to the present embodiment and an operation signalgenerator configured to generate operation signals when these operationelements are operated.

A display unit 14 includes a display device such as, for example, aliquid crystal display or an organic EL (ElectroLuminescent) display anda display driver circuit for the display. On the basis of an imagesignal supplied from an image processing unit 25, the display unit 14displays various characters such as, for example, an electronic mail anda message on the display and displays a still image, a video image, andthe like.

The image processing unit 25 performs a processing of generating imagesignals for the characters, symbols, images, etc., which are displayedon the display unit 14. Also, under the control of the control unit 10,the image processing unit 25 also displays various user interfacescreens, web pages, and the like.

The memory unit 15 includes a ROM (Read Only Memory) and a RAM (RandomAccess Memory). The ROM includes a rewritable storage medium such as aNAND-type flash memory, and also stores, for example, an OS (OperatingSystem) program, a control program for the control unit 10 to controlthe various units, various application program such as, for example, inaddition to compressed and encoded music data contents and video imagedata contents, various initial setting values, font data, respectivedictionary data, machine name information, and terminal identificationinformation. The RAM stores data as a work area for the control unit 10to perform various data processings as the occasion demands.

In addition, although not shown in FIG. 1, the mobile phone terminalaccording to the present embodiment is also provide with respectivecomponents provided to a general mobile phone terminal such as a digitalcamera unit for an image pickup of a photographic image, an LED (lightemitting diode) for key illumination, an incoming call light, and thelike, a drive unit for the LED, a battery for supplying electric powerto the respective units, a power management IC unit configured tocontrol the electric power of the battery, a close range wirelesscommunication unit configured to perform a close range wirelesscommunication based on so called Bluetooth (registered trademark)system, a UWB (Ultra Wide Band) system, a wireless LAN (Local AreaNetwork), a non-contact communication processing unit provided with anon-contact IC card function and a reader writer function, a GPS (GlobalPositioning System) communication unit, an external memory slot, adigital broadcasting reception tuner unit, an AV codec unit, and a timer(clock unit).

Description for Details of the Noise Detection Unit and the IncomingCall Audio Processing Unit and Operations

FIG. 2 shows detail configurations of the noise detection unit 23 andthe incoming call audio processing unit 24 provided in the audioprocessing unit 20 of the mobile phone terminal according to the presentembodiment. It should be noted that hereinafter, for convenience of thedescription, the example has been described in which the analog audiosignal is processed, but the embodiment of the present invention canalso of course be applied to a case in which a digitalized audio signalis processed.

In the mobile phone terminal according to the present embodiment, theincoming call audio processing unit 24 includes a dynamics adjustmentunit 50 and a formant adjustment unit 40. The dynamics adjustment unit50 is an input and output characteristic change function unit configuredto when the phone call is carried out by the mobile phone terminalaccording to the present embodiment, control the input and outputcharacteristics (dynamics) of the incoming call audio signal sent fromthe other party of the phone call in accordance with the control signalfrom the noise detection unit 23. The formant adjustment unit 40 is alevel adjustment function unit configured to perform a processing ofemphasizing the second formant which is in particular hardly overlappedwith the peak of the ambient environment noise among formants includedin the incoming call audio signal which will be described below. Thatis, the formant adjustment unit 40 functions as an equalizer configuredto lift up a contour component of the voice on the incoming call.

The noise detection unit 23 is composed of an ambient environment noisedetection unit and a control signal generation unit. When the phone callis carried out by the mobile phone terminal according to the presentembodiment, the noise detection unit 23 detects the ambient environmentnoise level included in the audio signal collected by the microphone 22and generates a control signal for controlling the input and outputcharacteristics of the dynamics adjustment unit 50 from the ambientenvironment noise level.

That is, the mobile phone terminal according to the present embodimentadjusts the level of the incoming call audio signal upon the audio phonecall on the basis of the ambient environment noise level. To be morespecific, for example, when the ambient environment noise level islarge, by controlling the dynamics on the audio reception side so as toincrease the level of the incoming call audio signal, it is possible tomake it easier to hear the phone call audio (in particular, the voice onthe incoming call) with a small processing amount.

In addition, in the mobile phone terminal according to the presentembodiment, the noise detection unit 23 also detects, for example, theambient environment noise caused by the voice of the other people exceptfor the voice of the talker during the phone call, and generates thecontrol signal on the basis of the ambient environment noise caused bythe voice of the other people.

With this configuration, in the mobile phone terminal according to thepresent embodiment, even in a case where the ambient environment noisecaused by the voice of the other people exists, it is possible to makeit easier to hear the incoming call audio.

Hereinafter, the above-mentioned configuration for making it easier tohear the incoming call audio and the operation will be specificallydescribed.

In FIG. 2, an audio signal output from the microphone 22 used for thephone call is input to a microphone audio input terminal 31 of the noisedetection unit 23 provided in the audio processing unit 20. The audiosignal input to the microphone audio input terminal 31 is amplified byan amplifier 32 and thereafter output as an outgoing call audio signalfrom an outgoing call audio output terminal 35 to a general circuit unitfor an outgoing call audio processing.

In addition, according to the present embodiment, the audio signalamplified by the amplifier 32 is branched from an output path for theabove-mentioned outgoing call audio and introduced into a low-passfilter (LPF) 33 and a band-pass filter 61 provided in parallel with eachother.

The respective branch paths to the low-pass filter 33 and the band-passfilter 61 are provided for examining how much the ambient environmentnoise is contained in the audio signal input from the microphone 22. Inthe case of the present embodiment, although a detail will be describedbelow, the branch path on the low-pass filter 33 side is provided forexamining the ambient environment noise from which the human voice issubstantially removed. On the other hand, the branch path on theband-pass filter 61 side is provided for examining the ambientenvironment noise caused by the voice of the other people except for thevoice of the talker during the phone call.

Herein, in the frequency characteristic of the human voice, as shown ina characteristic curve represented by the solid line in FIG. 3,particular peaks (formants) exist. Although an individual variationexists, the frequency of the formant has large two peaks (formants)about between 300 Hz to 3.4 kHz. The first formant exists in thevicinity of 500 kHz to 1 kHz, and the second formant exists in thevicinity of 1.5 kHz to 3 kHz.

On the other hand, variations of the ambient environment noise areconsiderable depending on an environment, but the frequencycharacteristic of the ambient environment noise in the general useenvironment for the mobile phone terminal often attenuates from a lowerband towards a higher hand as shown in a characteristic curverepresented by the dotted line in FIG. 3.

It should be noted that FIG. 4 is an amplitude-frequency characteristicdiagram of an actually measured incoming call audio signal including theambient environment noise, and FIG. 5 is an amplitude-frequencycharacteristic diagram of the incoming call audio signal of FIG. 4 inwhich 0 Hz to 300 Hz are expanded. Also, FIG. 6 is anamplitude-frequency characteristic diagram of a formant of the actuallymeasured incoming call audio signal, and FIG. 7 is anamplitude-frequency characteristic diagram of the incoming call audiosignal of FIG. 4 in which 0 Hz to 300 Hz are expanded.

In view of the above, the mobile phone terminal according to the presentembodiment is provided with the low-pass filter 33 having thecharacteristics shown, for example, in FIGS. 8 and 9 for identifying howmuch the ambient environment noise from which the human voice issubstantially removed is contained in the input audio signal from themicrophone 22. It should be noted that FIG. 9 shows the expanded lowfrequency band part of FIG. 8 by changing the scale size of the x axisof FIG. 8 (frequency axis). That is, according to the presentembodiment, for the low-pass filter 33, a filter is used which isprovided with the relatively precipitous characteristic shown, forexample, in FIGS. 8 and 9 and in which, as in FIG. 3, a band lower thanthe first formant in the frequency characteristic of the human voice isset as a cutoff frequency (second frequency band of the embodiment ofthe present invention). It should be noted that according to the presentembodiment, for the low-pass filter 33, a filter is used which has, forexample, the cutoff frequency of 50 Hz to 140 Hz (in particular, about100 Hz in the example of FIGS. 8 and 9) and about a fourth-orderChebyshev characteristic.

According to the present embodiment, as the low band pass by thelow-pass filter 33 is carried out, as shown in FIG. 10, an audio signalwaveform Bf input from the microphone 22 is set as a post LPF passsignal waveform Alpf. That is, the post LPF pass signal waveform Alpf isa signal waveform obtained by removing band components where componentsof the human voice are extremely rare (that is, the signal component ofthe ambient environment noise from which the human voice issubstantially removed) from the output signal of the microphone 22.

The abovementioned signal passing through the low-pass filter 33 (thatis, the signal component of the ambient environment noise from which thehuman voice is substantially removed) is sent to an envelope detector 34as shown in FIG. 2.

In the envelope detector 34, by performing the envelope detection on thesignal having passed through the low-pass filter 33, to be morespecific, by averaging and sampling the post LPF pass signal waveformAlpf for every certain time interval, a signal representing an energyrough transient of the above-mentioned ambient environment noise isgenerated. That is, according to the present embodiment, the outputsignal of the envelope detector 34 is a signal representing the resultof the examination on how much the ambient environment noise from whichthe human voice is substantially removed is contained in the audiosignal input from the microphone 22. It should be noted that thetemporal frequency (cycle of averaging for every certain time interval)for detecting the above-mentioned energy transient is detected in theenvelope detector 34 is not limited herein, but the temporal frequencyis desirably set in accordance with the processing time unit used in theincoming call audio processing unit 24 in the later stage (for example,100 msec).

Incidentally, the above-mentioned ambient environment noise componentsextracted by the low-pass filter 33 are limited to noise components in apitch extent lower than the human voice. It should be noted that themobile phone terminal is used, for example, in an environment such as aparty venue or a pub. That is, in a case where the mobile phone terminalis used in such an environment, the voice of surrounding other peopleother than the talker having the phone call is also contained in theabove-mentioned ambient environment noise.

On the other hand, in a case where the phone call is carried out by themobile phone terminal under the environment where the human voicebecomes the noise source, in order to execute the above-mentionedprocessing for the adjustment on the level of the incoming call audiosignal for improving the ease of hearing the voice on the phone call,the voice on the phone call and the noise component due to the voice ofsurrounding other people are distinguished from each other. That is, onthe basis of the voice on the phone call, if the level adjustment foremphasizing the incoming call audio signal is performed, the soundquality of the incoming call voice is deteriorated. Thus, for example, aprocessing is performed for detecting the voice of other people as thenoise component in a period in which only the voice of surrounding otherpeople exists except for a period in which the voice on the phone callexists.

In view of the above, according to the present embodiment, the branchpath of the band-pass filter 61 in the noise detection unit 23 isprovided for identifying how much the ambient environment noise causedby the voice of the other people except for the voice of the talkerduring the phone call is contained in the audio signal input from themicrophone 22.

The band-pass filter 61 has a characteristic shown, for example, in FIG.11. That is, the band-pass filter 61 is a filter provided with acharacteristic of allowing pass of a relatively high band from about 200Hz to several kHz including the band of the human voice (first frequencyband according to the embodiment of the present invention) as shown inFIG. 11.

According to the present embodiment, as the band pass is performedthrough the band-pass filter 61, as shown in FIG. 12, the audio signalwaveform Bf input from the microphone 22 is set as a post BPF passsignal waveform Abpf. That is, the post BPF pass signal waveform Abpf isa signal waveform obtained by extracting the band component close to thehuman voice component (that is, the signal component including theambient environment noise due to the human voice) from the output signalof the microphone 22.

Herein, as described above, in a case where the frequency band componentclose to the human voice component is used for the detection of theambient environment noise, there is a problem that the voice componentof the talker itself during the phone call is also detected as theambient environment noise.

In order to solve this problem, the mobile phone terminal according tothe present embodiment is provided with a mute control circuit 64 and amute circuit 63 in the branch path on the band-pass filter 61 side asshown in FIG. 2.

That is, the voice of the talker during the phone call and the voice ofother people functioning as the ambient environment noise are thought tohave substantially the same voice frequency bands but to have adifference in temporal continuity. To be more specific, as words areused during the phone call, the voice of the talker has less temporalcontinuity due to a break between sentences, intake of breath, call fromthe other party on the phone call, and the like and changes unsteadily.In contrast, the voice of other people functioning as the ambientenvironment noise becomes more stationary with fewer temporal breaks asthe number of other people is increased and the noise becomes higher.

The mute control circuit 64 and the mute circuit 63 are provided todistinguish the audios from the difference in the above-mentionedtemporal continuity of the voices.

While referring back to the description of FIG. 2, the signal passingthrough the band-pass filter 61 (the signal component of the human voiceband) is input via an envelope detector 62 similar to the envelopedetector 34 described above to the mute circuit 63 and the mute controlcircuit 64.

In the initial state, the mute circuit 63 is set to mute the inputsignal (mute ON), and when a mute OFF signal is supplied from the mutecontrol circuit 64, the above-mentioned mute is cancelled (mute OFF).

When the signal level from the envelope detector 62 exceeds apredetermined threshold, and the state carries on for a predeterminedperiod of time (about several seconds), the mute control circuit 64outputs the above-mentioned mute signal to the mute circuit 63. That is,the mute control circuit 64 measures the continuous time in a state inwhich the output signal level of the envelope detector 62 exceeds theabove-mentioned threshold, and when the states carries on for theabove-mentioned predetermined period of time, the mute control circuit64 outputs the mute OFF signal.

Also, after the mute of the mute circuit 63 is cancelled (after a mutecancellation signal is output), when the output signal level from theenvelope detector 62 dips from the above-mentioned predeterminedthreshold, the mute control circuit 64 outputs a mute ON signal forimmediately enabling the mute of the mute circuit 63 (mute ON).

That is, according to the present embodiment, in the branch path of theband-pass filter 61, in the case where the input signal to the mutecontrol circuit 64 is the non-stationary signal without the temporalcontinuity, that is, the signal corresponding to the audio signal of thetalker on the phone call, the mute circuit 63 is put into the mute ONstate, and the output signal of the envelope detector 62 is not outputto the later stage of the mute circuit 63. On the other hand, the inputsignal to the mute control circuit 64 is the stationary signal with thetemporal continuity, that is, the signal corresponding to the voice ofother people functioning as the ambient environment noise, the mutecircuit 63 is put into the mute OFF state, and the output signal of theenvelope detector 62 is output to the later stage of the mute circuit63.

The output signal of the mute circuit 63 is sent to a comparator 66. Inaddition, the output signal of the envelope detector 34 on the branchpath on the low-pass filter 33 side is also supplied to the comparator66. It should be noted that a mixing volume circuit 65 is providedbetween the mute circuit 63 and the comparator 66. The mixing volumecircuit 65 is provided for adjusting the balance of signal intensitieson both the branch path on the low-pass filter 33 side and the branchpath on the band-pass filter 61 side.

The comparator 66 compares the levels between the output signal of theenvelope detector 62 via the mute circuit 63 in the branch path on theband-pass filter 61 side and the output signal of the envelope detector34 in the branch path on the low-pass filter 33 side described above andoutputs the higher signal to the incoming call audio processing unit 24in a later stage as the control signal. It should be noted that thelevel comparison in the comparator 66 is carried out, for example, insynchronization with the cycle of the above-mentioned certain timeinterval in the envelope detector 34 and the envelope detector 62.

FIG. 13A shows an example of the output signal waveform of the envelopedetector 62, FIG. 13B shows an example of the output signal waveform ofthe mute circuit 63, and FIG. 13C shows an example of the output signalwaveform of the mixing volume circuit 65.

As shown in the examples of FIGS. 13A to 13C, in a case where inputsignals to the mute circuit 63 and the mute control circuit 64 arewaveform signals shown in FIG. 13A, when a state in which the level ofthe signal waveform exceeds a predetermined threshold Lth of FIG. 13Acontinues for a predetermined time Tth, the mute control circuit 64outputs the mute OFF signal to the mute circuit 63. With thisconfiguration, a waveform signal shown in FIG. 13B is output from themute circuit 63 in the mute OFF state.

Also, after that, in a case where the input signal to the mute controlcircuit 64 dips from the above-mentioned predetermined threshold Tth,the mute control circuit 64 immediately outputs the mute ON signal tothe mute circuit 63. With this configuration, the signal waveform is notoutput from the mute circuit 63 in the mute ON state as shown in FIG.13B.

Also, FIG. 14A shows an example of the output signal waveform of theenvelope detector 34 on the branch path side of the low-pass filter 33,FIG. 14B shows an example of the output signal waveform of the mixingvolume circuit 65, and FIG. 14C shows an example of the output signalwaveform (control signal) of the comparator 66.

As shown in the examples of FIGS. 14A to 14C, a the result of the levelcomparison between the output signal from the mixing volume circuit 66on the branch path on the band-pass filter 61 side and the output signalof the envelope detector 34 on the branch path on the low-pass filter 33side, the higher signal is output from the comparator 66.

As described above, according to the present embodiment, among thecontrol signal based on the ambient environment noise from which thehuman voice is substantially removed in the branch path on the low-passfilter 33 side and the control signal based on the ambient environmentnoise due to the voice of other people in the branch path on theband-pass filter 61 side, the signal having the higher signal level isoutput to the incoming call audio processing unit 24.

With this configuration, for example, in a case where the ambientenvironment noise due to the human voice is larger than the ambientenvironment noise from which the human voice is substantially removed,in the incoming call audio processing unit 24, the level of the incomingcall audio signal adjustment is carried out on the basis of the controlsignal in accordance with the ambient environment noise level due to thehuman voice. On the other hand, in a case where the ambient environmentnoise from which the human voice is substantially removed is larger thanthe ambient environment noise due to the human voice, in the incomingcall audio processing unit 24, the level of the incoming call audiosignal adjustment is carried out on the basis of the control signal inaccordance with the ambient environment noise level for which the humanvoice is substantially removed.

In addition, according to the present embodiment, the period in whichthe voice of the talker on the phone call exists and the period in whichonly the ambient environment noise due to the voice of other peopleexists can be clearly distinguished from each other. Thus, inparticular, in the period in which the voice of the talker on the phonecall exists, on the basis of the control signal in accordance with theambient environment noise level from which the human voice issubstantially removed, the level of the incoming call audio signaladjustment is performed in the incoming call audio processing unit 24.

In addition, according to the present embodiment, the ambientenvironment noise from which the human voice is substantially removedand the ambient environment noise due to the human voice can be detectedin parallel. Then, the control signal based on the ambient environmentnoise due to the human voice is generated in the branch path on theband-pass filter 61 side while being delayed by at least a predeterminedtime, and on the other side, the control signal based on the ambientenvironment noise from which the human voice is substantially removed isregularly generate in the branch path on the low-pass filter 33 side.That is, in the incoming call audio processing unit 24 according to thepresent embodiment, while utilizing the promptness of the response speedat the time of the level adjustment based on the control signalsregularly generated from the ambient environment noise from which thehuman voice is substantially removed, it is possible to perform thelevel adjustment on the basis of the ambient environment noise due tothe human voice.

Description of a Configuration and an Operation of the DynamicsAdjustment Unit

Hereinafter, detailed configurations and operations of the dynamicsadjustment unit 50 and the formant adjustment unit 40 in the incomingcall audio processing unit 24 will be described.

First, the dynamics adjustment unit 50 will be described, and thereafterthe formant adjustment unit 40 will be described.

In the incoming call audio processing unit 24, the incoming call audiosignal sent from the circuit unit for the normal incoming call audioprocessing (not shown) is input to an incoming call audio input terminal45.

This incoming call audio signal is sent to a band-pass filter (BPF) 44which will be described below of the formant adjustment unit 40 and alsoto a delay phase shifter unit 47.

The incoming call audio signal via the delay phase shifter unit 47 whichwill be described below and also via an adder 46 which will be describedbelow is amplified in an amplifier 48 of the dynamics adjustment unit 50as the occasion demands and then input to an auto level controller (ALC)49.

Also, the control signal output from the comparator 66 of the noisedetection unit 23 is subjected to the level limit for a part exceeding aregulated level by a limiter 51 of the dynamics adjustment unit 50 andis further subjected to the level adjustment by an amplifier 52 as theoccasion demands to be thereafter sent to the auto level controller 49.It should be noted that in a case where the signal waveform of thecontrol signal output from the comparator 66 is a waveform shown, forexample, in FIG. 15, the control signal waveform subjected to the levellimit by the limiter 51 of the dynamics adjustment unit 50 and the leveladjustment by the amplifier 52 becomes a waveform shown, for example, inFIG. 16.

The output signal of the auto level controller 49 is output to via anincoming call audio output terminal 53 to the speaker 21 for theincoming call. It should be noted that a detailed description of theauto level controller 49 according to the present embodiment will bedescribed below.

Herein, the general auto level controller (ALC) is configured to have acharacteristic in which the input and output characteristic curve isdecided as one curve, and the input level and the output levelcorrespond to one to one. In contrast to this, the auto level controller49 provided to the incoming call audio processing unit 24 according tothe present embodiment is configured to be able to change the input andoutput characteristics themselves on the basis of the control signalshown in FIG. 16. To be more specific, the auto level controller 49according to the present embodiment is configured to have the input andoutput characteristics having a variable hinge point as shown in FIGS.17 to 19. It should be noted that FIG. 18 is a diagram expanding thevicinity of the variable hinge point of FIG. 17. Also, FIG. 19 shows arelation between the level change of the control signal and the changeof the variable hinge point.

That is, as shown in FIGS. 17 and 18, for example, the auto levelcontroller 49 according to the present embodiment can change the valueof the output level with respect to the input level in a plurality ofstages, for example, every 1 dB up to maximum 10 dB (for example, 11stages for every step of 1 dB) within a predetermined input level rangeconsidered to be the signal level of the incoming call voice due to thehuman voice (in FIGS. 17 and 18, for example, in a range between −30 dBor higher and the upper limit of −10 dB). As shown in FIG. 19, as thevalue of the next control signal is higher with respect to the value ofthe control signal one before, the above-mentioned variable hinge pointis shifted by one stage in a direction in which the output level becomeshigher (1 rank up). On the other hand, the value of the next controlsignal is lower with respect to the value of the control signal onebefore, the level control is carried out so that the above-mentionedvariable hinge point is shifted in a direction in which the output levelbecomes lower (1 rank down).

To be more specific, in a case where the input level is within thepredetermined input level, for example, the value of the control signalis large (that is, in a case where the surrounding environment noise islarge), the auto level controller 49 according to the present embodimentperforms the dynamics control to change the above-mentioned variablehinge point in a direction in which the output level with respect to theinput level increased so that the effects of the auto level controllerare enhanced. On the other hand, for example, in a case where the valueof the control signal is small (that is, in a case where the surroundingenvironment noise is small), the auto level controller 49 according tothe present embodiment performs the dynamics control to change theabove-mentioned variable hinge point in a direction in which the outputlevel with respect to the input level approach to the one to onerelation so that the effects of the auto level controller aresuppressed.

In other words, in a case where the incoming call audio signal at acertain level or higher is input, the above-mentioned surroundingenvironment noise is large (that is, when the value of the controlsignal is large), the auto level controller 49 according to the presentembodiment adjusts the input and output characteristics of the autolevel controller, for example, in a direction in which the inputlevel:the output level=1:n is established (n in this case is a valuehigher than 1 and corresponding to the respective variable hinge pointsfor every 1 dB step described above). Thus, for example, as shown inFIG. 20, the output level of the incoming call audio signal isincreased, and the hearing of the incoming call audio is facilitated. Onthe other hand, when the surrounding environment noise is small (thatis, when the value of the control signal is small), the input and outputcharacteristics of the auto level controller is adjusted in a directionapproaching the input level:the output level=1:1. Thus, the soundquality deterioration of the incoming call voice on the basis of thedynamics control in the auto level controller is suppressed to minimum.It should be noted that the solid line in FIG. 20 represents theamplitude waveform of the actual incoming call audio signal. Thedashed-dotted line in FIG. 20 represents the amplitude waveform of theincoming call audio signal when the output level according to thepresent embodiment is increased.

As described above, according to the present embodiment, for example, ina case where the surrounding environment noise is increased, and theinput and output characteristics of the auto level controller 49 areadjusted in a direction approaching the input level:the outputlevel=1:n. Even when some sound quality deterioration is caused in theincoming call audio, the level of the incoming call voice becomes higherrelatively with respect to the surrounding environment noise, theincoming call audio becomes easier to hear. On the other hand, in a casewhere the surrounding environment noise is smaller and the input andoutput characteristics of the auto level controller 49 is adjusted in adirection of approaching the input level:the output level=1:1, as thelevel of the incoming call voice does not become higher but the level ofthe original surrounding environment noise is also low, the possibilityis lowered that the surrounding environment noise adversely affects thephone call. Also, as the sound quality deterioration of the incomingcall voice is lowered, the incoming call audio becomes easier to hear.

It should be noted that in the above description, as shown in FIGS. 17and 18, the variable hinge point has been described as an example inwhich the value of the output level with respect to the input level canbe changed in a plurality of stages, for example, for every 1 dB up tomaximum 10 dB. However, the variable hinge point is not only onechanging in a discontinuous manner but also may be one changing in acontinuous manner, for example.

Also, in the above-mentioned example, to simplify the description, thecase has been described in which the dynamics control based on thevariable hinge point in the auto level controller 49 is carried outwhile directly following the change in the size of the ambientenvironment noise. However, for example, in a case where the ambientenvironment noise is drastically changed, the incoming call audio afterthe above-mentioned dynamics control may drastically change in such amanner that the user feels sense of discomfort in the hearing. For thisreason, in order to avoid the above-mentioned drastic change, forexample, the dynamics control in the auto level controller 49 accordingto the present embodiment is designed to prepare a hysteresis at acertain level with respect to the change in the variable hinge point.

Also, in FIGS. 17 and 18 described above, the example has been describedin which for the characteristic curve at the part where the input andoutput characteristics of the auto level controller 49 are changed, thecharacteristic curve bending at the part of the certain predeterminedinput level (in the example of FIGS. 17 and 18, the input level of −20dB) (the characteristic curve of the variable hinge point) is used.However, for example, as shown in FIG. 21, it is also possible to usethe characteristic curve changing in a plurality of stages(discontinuous manner) or a continuous manner in a state in which a linesegment is provided while which is in parallel with the input level andthe output level curve corresponding to one to one and also has acertain length.

That is, in the case of the example of FIG. 21, the auto levelcontroller 49 can change the value of the output level with respect tothe input level for every 1 dB in a plurality of stages (for example, 11stages in units of 1 dB) up to, for example, 10 dB within thepredetermined input level range considered to be the signal level of theincoming call voice due to the human voice. When the value of the nextcontrol signal is higher with respect to the value of the control signalone before, the dynamics control is performed so that the gain isshifted by one stage in a direction in which the output level becomeshigher. On the other hand, when the value of the next control signal islower with respect to the value of the control signal one before, thedynamics control is performed so that the gain is shifted in a directionin which the output level becomes lower. In the case of the example ofFIG. 21, as only the input and output characteristic curve which theauto level controller 49 originally has is shifted in parallel, thenumber of the changes in the circuit configuration is small, and it istherefore possible to easily realize the circuit at a low cost. Itshould be noted that at the time of the dynamics control, for example,the input and output level is detected, and also an attack time foradjusting the gain (a time when the gain is decreased) and a recoverytime (a time when the gain is increased) are prepared. It is desiredthat the attack time and the recovery time are adjusted in accordancewith the detection value of the above-mentioned input and output levelso that the change in the gain is not drastic.

Also, the case of the analog processing has been exemplified in theabove description. However, in addition, for example, in a case where adigital processing is used, a relation shown in, for example, FIGS. 22and 23 is established between the control signal and the variable hingepoint, and the certain time interval (for example, 100 msec interval).Each time the control signal is input, the comparison is performedbetween the value of the control signal corresponding to the variablehinge point at the relevant time and the above-mentioned control signal.When the value of the input control signal is higher, the variable hingepoint may be shifted by one stage in the direction in which the outputbecomes higher. On the other hand, when the value of the input controlsignal is lower, the variable hinge point may be shifted in thedirection in which the output becomes lower. With such a configuration,even in the case of using the digital processing, the drastic change inthe variable hinge point can be prevented.

According to the present embodiment, as the above-mentioned procedure isperformed, without increasing the processing amount, the ease of hearingthe voice on the phone call under the ambient environment noise can beimproved.

Description for the Configuration and Operation of the FormantAdjustment Unit

Next, the formant adjustment unit 40 of the incoming call audioprocessing unit 24 will be described.

The control signal output from the comparator 66 of the incoming callaudio processing unit 24 is subjected to the level limit by a limiter 41of the formant adjustment unit 40 for a part exceeding the regulatedlevel and is further subjected to the level adjustment by an amplifier42 as the occasion demands to be thereafter sent to an amplifier 43 as acontrol signal. It should be noted that in a case where the signalwaveform of the control signal output from the comparator 66 is, forexample, the above-mentioned waveform as shown in FIG. 15, the controlsignal waveform after the level limit by the limiter 41 of the formantadjustment unit 40 is applied, and the level adjustment is performed inthe amplifier 42 has a waveform shown, for example, in FIG. 24.

Also, the band-pass filter 44 to which the incoming call audio signalfrom the incoming call audio input terminal 45 is input is a filterprovided with a frequency characteristic shown, for example, in FIG. 25.That is, the band-pass filter 44 is a filter allowing only the frequencyband of the second formant which is hardly overlapped with the peak ofthe ambient environment noise, in particular, among the frequency bandsof the incoming call audio signal. It should be noted that FIG. 26 showsan amplitude-frequency measurement diagram obtained by actuallymeasuring the incoming call audio signal input from the incoming callaudio signal input terminal 45, and FIG. 27 shows an amplitude-frequencymeasurement diagram obtained by actually measuring the incoming callaudio signal after a band pass processing by the band-pass filter 44.

The incoming call audio signal in the frequency band of the secondformant passing through the band-pass filter 44 is input to theamplifier 43.

Herein, the amplifier 43 is composed of an amplifier having a relationof an amplification factor shown in FIG. 28 with respect to the controlsignal. With this configuration, in the amplifier 43, like thecharacteristic curve indicated by the dashed-dotted line in FIG. 29similarly as in FIG. 3 described above, with respect to the signal inthe frequency band of the second formant in the incoming call audiosignal, the gain adjustment processing (emphasis processing) of FIG. 28in accordance with the relation between the control signal and theamplification factor is carried out. It should be noted that FIG. 30shows an amplitude-frequency measurement diagram obtained by actuallymeasuring the incoming call audio signal after the gain adjustment bythe amplifier 43.

Then, the output signal of the amplifier 43 is sent to the adder 46.

Also, the adder 46 is supplied with the incoming call audio signal afterthe delay and phase adjustment by the delay phase shifter unit 47. Itshould be noted that the delay phase shifter unit 47 is installed forproviding a delay similar to the delay in the band-pass filter 44 of theformant adjustment unit 40, to the incoming call audio signal which isinput to the incoming call audio input terminal.

In the adder 46, the incoming call audio signal after the time and phaseadjustment by the delay phase shifter unit 47, the output signal of theamplifier 43 (that is, the signal on which the gain adjustment of thesecond formant is performed) is added. That is, the output signal of theadder 46 is a signal on which, as shown in FIG. 29 described above, theprocessing is performed for emphasizing the second formant whose band ishardly overlapped with the peak of the ambient environment noise, inparticular, among the formants included in the incoming call audiosignal. It should be noted that FIG. 31 shows an amplitude-frequencymeasurement diagram obtained by actually measuring the incoming callaudio signal after the addition processing by the adder 46.

Then, the signal output from the adder 46 is sent to the amplifier 48 ofthe above-mentioned dynamics adjustment unit 50.

Another Configuration Example of Noise Detection Unit

FIG. 32 shows another configuration example of the noise detection unit23 provided in the audio processing unit 20 of the mobile phone terminalaccording to the present embodiment. It should be noted that in FIG. 32,the same reference numerals are assigned to the same components as theabove-described respective components of FIG. 2, and a descriptionthereof will be omitted.

In this configuration example of FIG. 32, in the noise detection unit23, instead of the comparator 66 shown in FIG. 2, an adder 67 isprovided.

That is, in FIG. 32, the adder 67 is supplied with the output signal ofthe envelope detector 34 on the branch path on the low-pass filter 33side described above and the output signal of the mixing volume circuit66 on the branch path on the band-pass filter 61 side described above.

In the adder 67, the output signal of the envelope detector 62 via themute circuit 63 in the branch path on the band-pass filter 61 side andthe output signal of the envelope detector 34 in the branch path on thelow-pass filter 33 side are added.

Then, in this example of FIG. 32, an addition signal from the adder 67is output to the incoming call audio processing unit 24 in a later stageas the control signal.

According to this configuration example of FIG. 32, a signal obtained byadding the control signal based on the ambient environment noise fromwhich the human voice is substantially removed in the branch path on thelow-pass filter 33 side to the control signal based on the ambientenvironment noise due to the voice of other people in the branch path onthe band-pass filter 61 side is output as the control signal to theincoming call audio processing unit 24.

That is, according to this configuration example of FIG. 32, in a periodin which the voice of the talker on the phone call does not exists, in acase where the ambient environment noise due to the voice of otherpeople exists, the control signal is generated in which the ambientenvironment noise due to the voice of other people and the ambientenvironment noise from which the human voice is substantially removedare both taken into account.

Therefore, according to the example of FIG. 32, in the incoming callaudio processing unit 24, the adjustment is performed on the basis ofboth the ambient environment noise due to the voice of the other peopleand the ambient environment noise from which the human voice issubstantially removed, the level of the incoming call audio signal.

Of course, in the case of this example too, as described above, in theincoming call audio processing unit 24, while utilizing the promptnessof the level adjustment response speed based on the control signalsregularly generated from the ambient environment noise from which thehuman voice is substantially removed, the level adjustment by theambient environment noise due to the human voice can also be performed.

As described above, according to the present embodiment, the processingis performed for controlling the input and output characteristics(dynamics) of the incoming call audio signal sent from the other partyof the phone call by utilizing both the ambient environment noise fromwhich the human voice is substantially removed and the ambientenvironment noise caused by the voice of the surrounding other peopleexcept for the voice of the talker during the phone call.

Therefore, according to the present embodiment, even for the use undernot only the general various use environments but also, for example, theenvironment where the human voice becomes the noise source such as theparty venue or the pub, it is possible to make it easier to hear theincoming call audio.

Also, the processing for controlling the dynamics of the incoming callaudio signal in accordance with the size of the ambient environmentnoise can be realized at the extremely small processing amount onlyincluding the low-pass filter, the envelope detection, and the autolevel control.

Furthermore, according to the present embodiment, the dynamics controlis performed on the incoming call audio signal, and at the same time,the processing is performed for emphasizing the second formant whoseband is hardly overlapped with the peak of the ambient environmentnoise, in particular, among the formants included in the incoming callaudio signal (the processing for lifting up the contour components ofthe voice on the incoming call), so that it is possible to make it moreeasier to hear the incoming call audio.

It should be noted that the description according to the above-mentionedembodiment is an example of the present invention. For this reason, thepresent invention is not limited to the above-mentioned respectiveembodiments, and various modifications can of course be made inaccordance with the design and the like without departing from thetechnical idea according to the present invention.

For example, according to the above-mentioned embodiment, the mobileterminal such as the mobile phone terminal is exemplified, but thepresent invention can also be applied to a fixed-line communicationterminal such as a land line. In addition, the present invention canalso be applied to various mobile terminals such as, for example, a PDAprovided with a voice call function (Personal Digital Assistants).

The present application contains subject matter related to thatdisclosed in Japanese Priority Patent Application JP filed in the JapanPatent Office on Sep. 1, 2008, the entire content of which is herebyincorporated by reference.

It should be understood by those skilled in the art that variousmodifications, combinations, sub-combinations and alterations may occurdepending on design requirements and other factors insofar as they arewithin the scope of the appended claims or the equivalents thereof.

1. An audio signal processing apparatus comprising: a band signalcomponent extraction unit including a filter designed to extract asignal waveform in a predetermined frequency band from an audio signalinput by audio collection means configured to collect at least anoutgoing call audio upon an audio phone call, and a first envelopedetector configured to detect an envelope of a signal waveform afterpass of the filter and to output the signal waveform after the envelopedetection as signal components in the predetermined frequency band; astationary signal component extraction unit configured to extract atleast a stationary signal component from the signal components extractedby the band signal component extraction unit; a signal adjustment unitprovided with a level adjustment function of adjusting an output signallevel with respect to an input signal level and an input and outputcharacteristics change function for changing the input and outputcharacteristics on the basis of a control signal when performing thelevel adjustment in the level adjustment function, and configured to setan incoming call audio signal upon the audio phone call as the inputsignal; and a control signal generation unit configured to generate thecontrol signal for changing the input and output characteristics of thesignal adjustment unit by using at least the stationary signal componentextracted by the stationary signal component extraction unit wherein thestationary signal component extraction unit includes a mute unitconfigured to mute the signal waveform after the envelope detection anda mute control unit configured to cancel mute of the mute unit when asignal level of the signal waveform after the envelope detection exceedsa predetermined threshold and also this state carries on for apredetermined period of time and to enable the mute of the mute unitwhen the signal level dips from the predetermined threshold after themute cancellation, and the control signal generation unit uses at leastthe stationary signal component composed of a signal waveform outputfrom the mute unit to generate the control signal.
 2. The audio signalprocessing apparatus according to claim 1, wherein the band signalcomponent extraction unit extracts signal components in a firstfrequency band and signal components in a second frequency band as thesignal components in the predetermined frequency band, wherein thestationary signal component extraction unit extracts the stationarysignal component from the signal components in the first frequency bandamong the signal components in the predetermined frequency band, andwherein the control signal generation unit compares a signal level ofthe stationary signal component extracted from the signal components inthe first frequency band by the stationary signal component extractionunit with a signal level of the signal components in the secondfrequency band extracted by the band signal component extraction unitand uses the signal components at the higher one of the signal level togenerate the control signal.
 3. The audio signal processing apparatusaccording to claim 1, wherein the band signal component extraction unitextracts signal components in a first frequency band and signalcomponents in a second frequency band as the signal components in thepredetermined frequency band, wherein the stationary signal componentextraction unit extracts the stationary signal component from the signalcomponents in the first frequency band among the signal components inthe predetermined frequency band, and wherein the control signalgeneration unit uses signal components obtained by adding the stationarysignal component extracted from the signal components in the firstfrequency band by the stationary signal component extraction unit withthe signal components in the second frequency band extracted by the bandsignal component extraction unit to generate the control signal.
 4. Theaudio signal processing apparatus according to claim 2 or 3, wherein theband signal component extraction unit extracts signal components in aband of a human voice as the first frequency band and extracts signalcomponents in a frequency band other than the band of the human voice asthe second frequency band.
 5. The audio signal processing apparatusaccording to any one of claims 2 to 3, wherein the band signal componentextraction unit includes a first filter designed to extract a signalwaveform in the first frequency band from an input audio signal, asecond filter designed to extract a signal waveform in the secondfrequency band from the input audio signal, a first envelope detectorconfigured to detect an envelope of a signal waveform after pass of thefirst filter, and a second envelope detector configured to detect anenvelope of a signal waveform after pass of the second filter, outputs asignal waveform after the envelope detection by the first envelopedetector as the signal components in the first frequency band, andoutputs a signal waveform after the envelope detection by the secondenvelope detector as the signal components in the second frequency band,wherein the stationary signal component extraction unit includes themute unit configured to mute the signal waveform after the envelopedetection by the first envelope detector and the mute control unitconfigured to cancel mute of the mute unit when a signal level of thesignal waveform after the envelope detection exceeds the predeterminedthreshold and also this state carries on for a predetermined period oftime and to enable the mute of the mute unit when the signal level dipsfrom the predetermined threshold after the mute cancellation and outputsan output waveform of the mute unit as the stationary signal component,and wherein the control signal generation unit uses the stationarysignal component composed of a signal waveform output from the mute unitand the signal components in the second frequency band composed of thesignal waveform after the envelope detection by the second envelopedetector to generate the control signal.
 6. The audio signal processingapparatus according to claim 1, wherein when a value of the signal levelof the signal components is higher than a regulated value, the controlsignal generation unit generates the control signal for changing theinput and output characteristics of the signal adjustment unit from thesignal components in a direction in which the output signal levelbecomes higher with respect to the input signal level.
 7. The audiosignal processing apparatus according to claim 6, wherein when the valueof the signal level of the signal components is changed in a diminishingdirection, the control signal generation unit generates the controlsignal for changing the input and output characteristics of the signaladjustment unit from the signal components in a direction a relationbetween input signal level and the output signal level approaches to oneto one.
 8. The audio signal processing apparatus according to claim 6,wherein the signal adjustment unit changes the input and outputcharacteristics on the basis of the control signal stepwise orcontinuously.
 9. The audio signal processing apparatus according toclaim 6, wherein the signal adjustment unit allows a hysteresis in thechange of the input and output characteristics on the basis of thecontrol signal.
 10. The audio signal processing apparatus according toclaim 6, wherein the signal adjustment unit has a formant adjustmentfunction of emphasizing a formant component included in an incoming callaudio signal upon the audio phone call and sets an incoming call audiosignal after an adjustment processing on a predetermined formantcomponent through the formant adjustment function as the input signal.11. An audio signal processing method comprising the steps of:extracting a signal waveform in a predetermined frequency band by a bandsignal component extraction unit including a filter designed to extractthe signal waveform from an audio signal input by audio collection meansconfigured to collect at least an outgoing call audio upon an audiophone call and detecting an envelope of a signal waveform after pass ofthe filter by an envelope detector and outputting the signal waveformafter the envelope detection as the signal components in thepredetermined frequency band; extracting at least a stationary signalcomponent by a stationary signal component extraction unit from thesignal components extracted by the band signal component extractionunit; generating a control signal for changing input and outputcharacteristics of the signal adjustment unit by a control signalgeneration unit by using at least the stationary signal componentextracted by the stationary signal component extraction unit; setting anincoming call audio signal upon the audio phone call as an input signal,and by a signal adjustment unit on the basis of the control signalgenerated by the control signal generation unit, changing the input andoutput characteristics for adjusting an output signal level with respectto an input signal level to adjust a level of the incoming call audiosignal; muting the signal waveform after the envelope detection by amute unit of the stationary signal component extraction unit andcancelling mute of the mute unit by a mute control unit when a signallevel of the signal waveform after the envelope detection exceeds apredetermined threshold and also this state carries on for apredetermined period of time and enabling the mute of the mute unit whenthe signal level dips from the predetermined threshold after the mutecancellation; and using by the control signal generation unit at leastthe stationary signal component composed of a signal waveform outputfrom the mute unit to generate the control signal.
 12. A communicationterminal comprising: a communication unit configured to perform acommunication for at least an audio phone call; an audio collection unitconfigured to collect at least an outgoing call audio upon the audiophone call; an audio emission unit configured to convert an incomingcall audio signal upon the audio phone call into an acoustical wave tobe output; a band signal component extraction unit including a filterdesigned to extract a signal waveform in a predetermined frequency bandfrom the audio signal input by the audio collection unit, and anenvelope detector configured to detect an envelope of a signal waveformafter pass of the filter and to output the signal waveform after theenvelope detection as signal components in the predetermined frequencyband; a stationary signal component extraction unit configured toextract at least a stationary signal component from the signalcomponents extracted by the band signal component extraction unit; asignal adjustment unit provided with a level adjustment function ofadjusting an output signal level with respect to an input signal leveland an input and output characteristics change function for changing theinput and output characteristics on the basis of a control signal whenperforming the level adjustment in the level adjustment function, andconfigured to set the incoming call audio signal upon the audio phonecall as the input signal; and a control signal generation unitconfigured to generate the control signal for changing the input andoutput characteristics of the signal adjustment unit by using at leastthe stationary signal component extracted by the stationary signalcomponent extraction unit, wherein the stationary signal componentextraction unit includes a mute unit configured to mute the signalwaveform after the envelope detection and a mute control unit configuredto cancel mute of the mute unit when a signal level of the signalwaveform after the envelope detection exceeds a predetermined thresholdand also this state carries on for a predetermined period of time and toenable the mute of the mute unit when the signal level dips from thepredetermined threshold after the mute cancellation, the control signalgeneration unit uses at least the stationary signal component composedof a signal waveform output from the mute unit to generate the controlsignal, and the incoming call audio signal output from the signaladjustment unit is supplied to the audio emission unit.
 13. An audiosignal processing apparatus comprising: a band signal componentextraction unit including a filter designed to extract a signal waveformin a predetermined frequency band from an audio signal input by an audiocollection section configured to collect at least an outgoing call audioupon an audio phone call, and an envelope detector configured to detectan envelope of a signal waveform after pass of the filter and to outputthe signal waveform after the envelope detection as signal components inthe predetermined frequency band; a stationary signal componentextraction unit configured to extract at least a stationary signalcomponent from the signal components extracted by the band signalcomponent extraction unit; a signal adjustment unit provided with alevel adjustment function of adjusting an output signal level withrespect to an input signal level and an input and output characteristicschange function for changing the input and output characteristics on thebasis of a control signal when performing the level adjustment in thelevel adjustment function, and configured to set an incoming call audiosignal upon the audio phone call as the input signal; and a controlsignal generation unit configured to generate the control signal forchanging the input and output characteristics of the signal adjustmentunit by using at least the stationary signal component extracted by thestationary signal component extraction unit, wherein the stationarysignal component extraction unit includes a mute unit configured to mutethe signal waveform after the envelope detection and a mute control unitconfigured to cancel mute of the mute unit when a signal level of thesignal waveform after the envelope detection exceeds a predeterminedthreshold and also this state carries on for a predetermined period oftime and to enable the mute of the mute unit when the signal level dipsfrom the predetermined threshold after the mute cancellation, and thecontrol signal generation unit uses at least the stationary signalcomponent composed of a signal waveform output from the mute unit togenerate the control signal.